similar to: e164.org and tollfree ENUM records

Hi, I am having a strange problem with enum and AGI. Here is what happens: I have in my agi something like that: foreach my $resolver ("e164.arpa", "e164.info", "e164.org") { my @enums = get_enums($phone, $resolver); foreach my $enum (@enums) { $dialstring = $enum .

ENUM lookups at e164.org return a IP route for ALL toll-free numbers. I was surprised to observe that ALL toll-free numbers get a hit at e164.org. It appears that ALL toll-free prefixes have been delegated, thereby publishing an IP route for YOUR TOLL-FREE NUMBERS, my toll-free numbers, and even toll-free numbers that have not been allocated. :-) See below Should I care? Even though this

Dialing out If the AGI application dials outward by executing Dial, control over the call returns to the dialplan and the script loses contact with the Asterisk server. The script continues to run in the background by itself and is free to clean up and do post-dial processing. If you want your application to initiate a call out without being started through the dialplan: * Asterisk auto-dial

Dialing out If the AGI application dials outward by executing Dial, control over the call returns to the dialplan and the script loses contact with the Asterisk server. The script continues to run in the background by itself and is free to clean up and do post-dial processing. If you want your application to initiate a call out without being started through the dialplan: * Asterisk auto-dial

Does anyone have a better ENUM lookup handler than the built-in ENUMLOOKUP() function? The built-in function does not properly handle multiple return values such as: 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u" "E2U+SIP" "!^\\+1866(.*)$!sip:1866\\1 at tollfree.sip-happens.com!" . 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u"

Last night Mark and I made some changes to the IAXTEL tollfree outbound, and inbound access. The inbound access number has changed to: 248-724-0700. (This number is in Pontiac, MI Ratecenter, and is supplied by Telesthetic LLC, a next gen phone compnay) This number will say "Please dial your number now" at that point you can dial your 1-700-XXX-XXXX IAXTEL number assigned. In the

Hi, I recall seeing a sample extensions.conf file that allowed tollfree calls to be routed via iaxtel to the US and the NL, but I must be going blind, because I've scoured the list but can't find it. Can someone send it to me if they have it? Much appreciated. Thanks! --- Paul Cheng M?ty?s kir?ly ut 10 H-1121 Budapest HUNGARY paul.cheng@alum.mit.edu mobile: +36 30 381-9311

Hallo all do any of you know if the toll free access to the Netherlands is still working via FWD or Iaxtel? thanks liaan --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I've got a problem with connection to my SIP provider. In general, everything works, but I get lots of these messages: Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's odd... Got a response on a call we dont know about. Cseq 42710 Cmd SIP/2.0 Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's odd... Got a response on a call

Calling from my home using Asterisk 1.6.2.1 to an office extension (Asterisk 1.6.1.13) the callerid is not honored: Home: -- Starting simple switch on 'DAHDI/1-1' -- Executing [170 at internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [170 at internal:2] NoOp("DAHDI/1-1", "Context: office-extensions") in new stack

Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get "stuck" in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log

Hi, I'm having trouble explicitly forcing a codec between sip devices. Am I missing something or is this not really possible? I have a grandstream registering to asterisk, named sip0. Sip0 registers, via sip, to another asterisk box, sip1. When I place a call from the grandstream, it will travel through sip0 to sip1, where it is then placed to the PSTN. Nothing can reinvite, this path is

Hello list, my case is as follows: SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729. When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the keypad on the phone. As suggested by you, I need to configure the SIP1 with out band dtmf mode , what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238 ? do I also need to make same kind

I'm reposting this to the list.. My spam filters didn't like the list host. :( If anyone was able to respond to the mail below, can you send it again please? Thanks. ------------------------------------------------------------------------- Hi, I'm having trouble explicitly forcing a codec between sip devices. Am I missing something or is this not really possible? I have a

3rd attempt to post it to the list, please ignore if it is duplicate I have the following problem When trying to setup asterisk 13.1 with PJSIP to connect to my IPV6 capable SIP provider the registration fails. [code][Dec 22 19:24:24] DEBUG[25247] pjsip: tsx0x110736c .Transaction created for Request msg REGISTER/cseq=36181 (tdta0x721d90) [Dec 22 19:24:24] DEBUG[25247] pjsip:

I've setup a SIP channel between two Asterisk box, and use Manager API to generate some calls. It's working quite fine, except this message (on the caller-side) : Jan 10 18:18:09 WARNING[25046]: chan_sip.c:6805 handle_response: Forbidden - wrong password on authentication for INVITE to '"sip1" <sip:asterisk@192.168.1.200>;tag=as77e9ebbb' But the call is going

Grrr. I'm using outlook web access and there's no way to do inline replies. Anyway... Gabriel. Using SER does not create a single point of failure. You install three SER boxes. Single point of failure gone. It does not take several seconds. If your phones are configured for SRV, and 2/3 of your SER boxes down, it takes about 2s. That's not bad for a 2/3 system failure. You can

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

Hi I have just been struggling for four days to get SIP working and now as I created a voicemail box, Asterisk has become very unstable and it can't bridge SIP phone to SIP provider calls anymore. Calling internally from one SIP phone to another works fine. Calling internally from a SIP phone to an analog phone on a Zap channel and vice versa works fine. Incoming PSTN calls delivered to

Hello to all. I have a couple of budgetones connected to Asterisk server. I can establish calls using budgetone with no problem, but when I hang up a Budgetone, Asterisk does not detect the hangup. It seems that the communication goes on in spite of budgetone's hangup. My sip.conf: [general] disallow=all allow=ulaw bindaddr=172.16.60.21 [sip1] callgroup=1 pickupgroup=1 type=friend